Re: [asterisk-users] Everyone is busy/congested: IP Trunk
Well, unfortunately i did not dig much into "why/how it worked" with openvpn, but it did work for me with default setup.I think you may need to set constant ports instead of random ports. Thanks, Vivek On 11/9/07, bilal ghayyad <[EMAIL PROTECTED]> wrote: > > Hi Friends; > > Actually I would appreciate if Vivek can advise if the > VPN resolved the RTP packets in the SIP Trunk between > Asterisk and another softswitch? In other words, > openvpn helpful in NAT cases in what exactly? As > without VPN, I was able to establish a call but > without voice or with complete noise (nothing > understood) :) - So if NAT resolve this issue for the > SIP Trunk, then I can proceed forward, as really now I > do not have any other attempt to try. > > From the other side, I think that baji is talking > about something else than the IP Trunk, he is talking > about outbound (which is related to using an > application to run an outside call, which is used > usually in campaign in contact centers and so on), I > think nthis case differs that placing a calls via IP > Trunk or even outside call but the caller who will do > it (and not the application). > > Lastly, Mr. Amit helped me when he gave me a > configuration to be done for the SIP Trunk, as in his > method, I did not register on the softswitch, I send > directly, and the connectioned succeed, but as I said: > with complete voice (actually nothing understood, i > feel it is complete RTP situation), the test was by > letting Asterisk behind NAT (private IP) and sending > to a softswitch in anther country has a public IP > address. Is it NAT issue, so VPN can resolve? > > Note: anyone knows if h323 works better in the IP > trunk? > > Regards > Bilal > > -- > yeah i found openvpn helpful in NAT cases. > > -Vivek > > > On 11/6/07, Baji Panchumarti > <[EMAIL PROTECTED]> wrote: > > > > after a copious loss of follicles :-), I finally got > outbound > working. > > > > Basically the channel statement in the call file > needs to have the > > number to be called. For eg., in test.call format > the statement > > as follows : > > > >Channel: SIP/3012345678@ > > > > And there is no need for a DIAL statement in > extensions.conf > > unless you need to dial an additional number / > extension. > > > > Then in sip.conf you need a para that matches > > > with the relevant auth info. > > > > These two wiki pages, they were very helpful in > figuring out a > > solution to the problem : > > > > > http://www.voip-info.org/wiki/index.php?page=Asterisk+auto-dial+out > > > > > > > > > http://www.voip-info.org/wiki/index.php?page=Asterisk+auto-dial+out+deliver+message > > > > hth, > > > > -baji. > > > > -- > > > > On Oct 30, 2007 8:43 AM, Gabriel Natale wrote: > > > > > I have the same problem. > > > > > > I trying with more 4 SIP providers, the account is > registering, > receive > > > inboud calls, but can`t make outbound calls for > "congestion". > > > > > > Can be the out call id the problem? > > > > > > Thanks > > > Gabriel > > > __ > Do You Yahoo!? > Tired of spam? Yahoo! Mail has the best spam protection around > http://mail.yahoo.com > ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Everyone is busy/congested: IP Trunk
On Nov 9, 2007 2:53 PM, bilal ghayyad wrote: > [...] > > From the other side, I think that baji is talking about > something else than the IP Trunk, he is talking > about outbound [...] correct, I was responding to Gabriel's post on being registered w/ SIP provider & accepting inbound, but having trouble w/ outbound. thnx, -baji. -- ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Everyone is busy/congested: IP Trunk
Hi Friends; Actually I would appreciate if Vivek can advise if the VPN resolved the RTP packets in the SIP Trunk between Asterisk and another softswitch? In other words, openvpn helpful in NAT cases in what exactly? As without VPN, I was able to establish a call but without voice or with complete noise (nothing understood) :) - So if NAT resolve this issue for the SIP Trunk, then I can proceed forward, as really now I do not have any other attempt to try. >From the other side, I think that baji is talking about something else than the IP Trunk, he is talking about outbound (which is related to using an application to run an outside call, which is used usually in campaign in contact centers and so on), I think nthis case differs that placing a calls via IP Trunk or even outside call but the caller who will do it (and not the application). Lastly, Mr. Amit helped me when he gave me a configuration to be done for the SIP Trunk, as in his method, I did not register on the softswitch, I send directly, and the connectioned succeed, but as I said: with complete voice (actually nothing understood, i feel it is complete RTP situation), the test was by letting Asterisk behind NAT (private IP) and sending to a softswitch in anther country has a public IP address. Is it NAT issue, so VPN can resolve? Note: anyone knows if h323 works better in the IP trunk? Regards Bilal -- yeah i found openvpn helpful in NAT cases. -Vivek On 11/6/07, Baji Panchumarti <[EMAIL PROTECTED]> wrote: > > after a copious loss of follicles :-), I finally got outbound working. > > Basically the channel statement in the call file needs to have the > number to be called. For eg., in test.call format the statement > as follows : > >Channel: SIP/3012345678@ > > And there is no need for a DIAL statement in extensions.conf > unless you need to dial an additional number / extension. > > Then in sip.conf you need a para that matches > with the relevant auth info. > > These two wiki pages, they were very helpful in figuring out a > solution to the problem : > > http://www.voip-info.org/wiki/index.php?page=Asterisk+auto-dial+out > > > http://www.voip-info.org/wiki/index.php?page=Asterisk+auto-dial+out+deliver+message > > hth, > > -baji. > > -- > > On Oct 30, 2007 8:43 AM, Gabriel Natale wrote: > > > I have the same problem. > > > > I trying with more 4 SIP providers, the account is registering, receive > > inboud calls, but can`t make outbound calls for "congestion". > > > > Can be the out call id the problem? > > > > Thanks > > Gabriel __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Everyone is busy/congested: IP Trunk
yeah i found openvpn helpful in NAT cases. -Vivek On 11/6/07, Baji Panchumarti <[EMAIL PROTECTED]> wrote: > > after a copious loss of follicles :-), I finally got outbound working. > > Basically the channel statement in the call file needs to have the > number to be called. For eg., in test.call format the statement > as follows : > >Channel: SIP/3012345678@ > > And there is no need for a DIAL statement in extensions.conf > unless you need to dial an additional number / extension. > > Then in sip.conf you need a para that matches > with the relevant auth info. > > These two wiki pages, they were very helpful in figuring out a > solution to the problem : > > http://www.voip-info.org/wiki/index.php?page=Asterisk+auto-dial+out > > > http://www.voip-info.org/wiki/index.php?page=Asterisk+auto-dial+out+deliver+message > > hth, > > -baji. > > -- > > On Oct 30, 2007 8:43 AM, Gabriel Natale wrote: > > > I have the same problem. > > > > I trying with more 4 SIP providers, the account is registering, receive > > inboud calls, but can`t make outbound calls for "congestion". > > > > Can be the out call id the problem? > > > > Thanks > > Gabriel > > ___ > --Bandwidth and Colocation Provided by http://www.api-digital.com-- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Everyone is busy/congested: IP Trunk
after a copious loss of follicles :-), I finally got outbound working. Basically the channel statement in the call file needs to have the number to be called. For eg., in test.call format the statement as follows : Channel: SIP/3012345678@ And there is no need for a DIAL statement in extensions.conf unless you need to dial an additional number / extension. Then in sip.conf you need a para that matches with the relevant auth info. These two wiki pages, they were very helpful in figuring out a solution to the problem : http://www.voip-info.org/wiki/index.php?page=Asterisk+auto-dial+out http://www.voip-info.org/wiki/index.php?page=Asterisk+auto-dial+out+deliver+message hth, -baji. -- On Oct 30, 2007 8:43 AM, Gabriel Natale wrote: > I have the same problem. > > I trying with more 4 SIP providers, the account is registering, receive > inboud calls, but can`t make outbound calls for "congestion". > > Can be the out call id the problem? > > Thanks > Gabriel ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Everyone is busy/congested: IP Trunk
Dear Amit;
Special thanks for your greate help and support.
Sorry for delaying in reply, I was busy during this
week.
It worked with very poor and noise voice, and
disconnect after around 5 seconds, but it worked in
the direct mode (by using trustip=yes so Asterisk does
not register on the softswitch). But maybe this voice
problem was because of the network (I will explain my
network situation).
Before I explain my network situation, I would like to
know why in registering mode (by using register =>
directive and letting asterisk registering on the
softswitch), Asterisk was registering successfully but
call was not arrive for the softswitch (does not know
if Asterisk sent it or did not send it). The question:
is there a kind of packets negotiation during the SIP
registeration that determine the facility of call
exchaning? The softswitch ables to receive calls from
any SIP endpoint, why this does not do happen with
Asterisk if Asterisk registered? But it receive and
manipulate the calls if Asterisk work via trustip
(without registeration)?!! Actually, when Asterisk was
registering on the softswitch, I was see the
registeration on the softswitch, but I did not see
even the call attempt.
Regarding to my network status (that might be the
reason of having very poor and noise voice and
disconnecting the line after around 5 second),
actually the softswitch in public IP address and it is
located in Germany, while the Asterisk in Kuwait and
it is behind NAT (a private IP address), and the
softphone also have a private IP address (in the same
LAN with the Asterisk), so the softphone was
registering on the Asterisk, when the softphone send
the call for Asterisk then Asterisk was sending it for
the the softswitch in Germany via the SIP Trunk.
Do u think that because Asterisk Nated? In that case,
do u think the VPN will resolve the problem (VPN
between Asterisk network in Kuwait, and the Softswitch
network in Germany)? Or there is a settings should be
done?
Regards
Bilal
I have the same problem.
I trying with more 4 SIP providers, the account is
registering, receive
inboud calls, but can`t make outbound calls for
"congestion".
Can be the out call id the problem?
Thanks
Gabriel
- Original Message -
From: <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Commercial
Discussion"
Sent: Monday, October 29, 2007 6:54 PM
Subject: Re: [asterisk-users] Everyone is
busy/congested: IP Trunk
> No:
>
> register => abc:[EMAIL PROTECTED]
>
> [peer]
> host=zzz
>
> Its possible to make mistakes and typos you know.
Maybe you can post
> your config file and we can help you.
>
> On 10/26/07, bilal ghayyad <[EMAIL PROTECTED]>
wrote:
>> Hi Pablo;
>>
>> How the IP address will be wrong, and asterisk able
to
>> do registeration on the destination?
>>
>> If the IP address wrong, so I will not be able to
>> register on that IP address.
>>
>> Regards
>> Bilal
>>
>> > Hi List;
>>
>>
>> Ip address to destination?
>>
>> Unable to create channel of type SIP (cause 3 - No
>> route to destination)
>>
>> i think you have the wrong ip information
>>
>>
>>
>> >
>> > I established an SIP IP Trunk between Asterisk
and
>> > another softswitch (asterisk registered on the
>> > softswitch successfully) and I saw this on the
>> > softswitch.
>> >
>> > >From firefly softphone, I was need to do a call
to
>> be
>> > via this softswitch (ofcourse, the softphone will
>> send
>> > for asterisk and asterisk should route to the
>> > softswitch based on the extensions.conf
>> > configurations.
>> >
>> > But, always I receive this message (and the call
>> does
>> > not even reach to the softswitch, it is not
sended
>> > from Asterisk to the softswitch):
>> >
>> > Executing [EMAIL PROTECTED]:1]
>> > Dial("SIP/EgyptOeratorSIP-09f9bed0",
>> > "SIP/[EMAIL PROTECTED]") is new stack
>> >
>> > Unable to create channel of type SIP (cause 3 -
No
>> > route to destination)
>> >
>> > Everyone is busy/congested at this time (1:0/0/1)
>> >
>> > Anyone faced that?
>> >
>> > Is it related to a paramater that control number
of
>> > allowed channels per IP trunk? Maybe I have such
>> > parameters is 0 ? I do not know even if there is
>> such
>> > parameter.
>> >
>> > At the softswitch, I do not see even any attempt
>> > (nothing related to the dialed number), so why
>> > Asterisk does not send the called number to the
>> > softswitch and why asterisk a
Re: [asterisk-users] Everyone is busy/congested: IP Trunk
I have the same problem.
I trying with more 4 SIP providers, the account is registering, receive
inboud calls, but can`t make outbound calls for "congestion".
Can be the out call id the problem?
Thanks
Gabriel
- Original Message -
From: <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
Sent: Monday, October 29, 2007 6:54 PM
Subject: Re: [asterisk-users] Everyone is busy/congested: IP Trunk
> No:
>
> register => abc:[EMAIL PROTECTED]
>
> [peer]
> host=zzz
>
> Its possible to make mistakes and typos you know. Maybe you can post
> your config file and we can help you.
>
> On 10/26/07, bilal ghayyad <[EMAIL PROTECTED]> wrote:
>> Hi Pablo;
>>
>> How the IP address will be wrong, and asterisk able to
>> do registeration on the destination?
>>
>> If the IP address wrong, so I will not be able to
>> register on that IP address.
>>
>> Regards
>> Bilal
>>
>> > Hi List;
>>
>>
>> Ip address to destination?
>>
>> Unable to create channel of type SIP (cause 3 - No
>> route to destination)
>>
>> i think you have the wrong ip information
>>
>>
>>
>> >
>> > I established an SIP IP Trunk between Asterisk and
>> > another softswitch (asterisk registered on the
>> > softswitch successfully) and I saw this on the
>> > softswitch.
>> >
>> > >From firefly softphone, I was need to do a call to
>> be
>> > via this softswitch (ofcourse, the softphone will
>> send
>> > for asterisk and asterisk should route to the
>> > softswitch based on the extensions.conf
>> > configurations.
>> >
>> > But, always I receive this message (and the call
>> does
>> > not even reach to the softswitch, it is not sended
>> > from Asterisk to the softswitch):
>> >
>> > Executing [EMAIL PROTECTED]:1]
>> > Dial("SIP/EgyptOeratorSIP-09f9bed0",
>> > "SIP/[EMAIL PROTECTED]") is new stack
>> >
>> > Unable to create channel of type SIP (cause 3 - No
>> > route to destination)
>> >
>> > Everyone is busy/congested at this time (1:0/0/1)
>> >
>> > Anyone faced that?
>> >
>> > Is it related to a paramater that control number of
>> > allowed channels per IP trunk? Maybe I have such
>> > parameters is 0 ? I do not know even if there is
>> such
>> > parameter.
>> >
>> > At the softswitch, I do not see even any attempt
>> > (nothing related to the dialed number), so why
>> > Asterisk does not send the called number to the
>> > softswitch and why asterisk assume there is not
>> > available channel?
>> >
>> > The softphone codec is g729a and the softswitch
>> > support such codec. Also, if it is a codec matter,
>> > then call should be send to the softswitch, and the
>> > softswitch will gives an error related to the codec
>> > missmatch.
>> >
>> > Any help?
>> >
>> > Regards
>> > Bilal Ghayad
>>
>>
>> __
>> Do You Yahoo!?
>> Tired of spam? Yahoo! Mail has the best spam protection around
>> http://mail.yahoo.com
>>
>> ___
>> --Bandwidth and Colocation Provided by http://www.api-digital.com--
>>
>> asterisk-users mailing list
>> To UNSUBSCRIBE or update options visit:
>>http://lists.digium.com/mailman/listinfo/asterisk-users
>>
>
> ___
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>
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> To UNSUBSCRIBE or update options visit:
> http://lists.digium.com/mailman/listinfo/asterisk-users
>
>
>
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Re: [asterisk-users] Everyone is busy/congested: IP Trunk
No:
register => abc:[EMAIL PROTECTED]
[peer]
host=zzz
Its possible to make mistakes and typos you know. Maybe you can post
your config file and we can help you.
On 10/26/07, bilal ghayyad <[EMAIL PROTECTED]> wrote:
> Hi Pablo;
>
> How the IP address will be wrong, and asterisk able to
> do registeration on the destination?
>
> If the IP address wrong, so I will not be able to
> register on that IP address.
>
> Regards
> Bilal
>
> > Hi List;
>
>
> Ip address to destination?
>
> Unable to create channel of type SIP (cause 3 - No
> route to destination)
>
> i think you have the wrong ip information
>
>
>
> >
> > I established an SIP IP Trunk between Asterisk and
> > another softswitch (asterisk registered on the
> > softswitch successfully) and I saw this on the
> > softswitch.
> >
> > >From firefly softphone, I was need to do a call to
> be
> > via this softswitch (ofcourse, the softphone will
> send
> > for asterisk and asterisk should route to the
> > softswitch based on the extensions.conf
> > configurations.
> >
> > But, always I receive this message (and the call
> does
> > not even reach to the softswitch, it is not sended
> > from Asterisk to the softswitch):
> >
> > Executing [EMAIL PROTECTED]:1]
> > Dial("SIP/EgyptOeratorSIP-09f9bed0",
> > "SIP/[EMAIL PROTECTED]") is new stack
> >
> > Unable to create channel of type SIP (cause 3 - No
> > route to destination)
> >
> > Everyone is busy/congested at this time (1:0/0/1)
> >
> > Anyone faced that?
> >
> > Is it related to a paramater that control number of
> > allowed channels per IP trunk? Maybe I have such
> > parameters is 0 ? I do not know even if there is
> such
> > parameter.
> >
> > At the softswitch, I do not see even any attempt
> > (nothing related to the dialed number), so why
> > Asterisk does not send the called number to the
> > softswitch and why asterisk assume there is not
> > available channel?
> >
> > The softphone codec is g729a and the softswitch
> > support such codec. Also, if it is a codec matter,
> > then call should be send to the softswitch, and the
> > softswitch will gives an error related to the codec
> > missmatch.
> >
> > Any help?
> >
> > Regards
> > Bilal Ghayad
>
>
> __
> Do You Yahoo!?
> Tired of spam? Yahoo! Mail has the best spam protection around
> http://mail.yahoo.com
>
> ___
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>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users
>
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Re: [asterisk-users] Everyone is busy/congested: IP Trunk
On 10/27/07, bilal ghayyad <[EMAIL PROTECTED]> wrote: > > Hi Pablo; > > How the IP address will be wrong, and asterisk able to > do registeration on the destination? > > If the IP address wrong, so I will not be able to > register on that IP address. Hi i see 2 causes 1. it could be Dialplan issue ( check how the provider accept the call, 1 or just USA number) 2 provider blocked account check network trace to get more info ngrep should be the ideal tool to check the errors in network trace ram ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Everyone is busy/congested: IP Trunk
Hi Pablo;
How the IP address will be wrong, and asterisk able to
do registeration on the destination?
If the IP address wrong, so I will not be able to
register on that IP address.
Regards
Bilal
> Hi List;
Ip address to destination?
Unable to create channel of type SIP (cause 3 - No
route to destination)
i think you have the wrong ip information
>
> I established an SIP IP Trunk between Asterisk and
> another softswitch (asterisk registered on the
> softswitch successfully) and I saw this on the
> softswitch.
>
> >From firefly softphone, I was need to do a call to
be
> via this softswitch (ofcourse, the softphone will
send
> for asterisk and asterisk should route to the
> softswitch based on the extensions.conf
> configurations.
>
> But, always I receive this message (and the call
does
> not even reach to the softswitch, it is not sended
> from Asterisk to the softswitch):
>
> Executing [EMAIL PROTECTED]:1]
> Dial("SIP/EgyptOeratorSIP-09f9bed0",
> "SIP/[EMAIL PROTECTED]") is new stack
>
> Unable to create channel of type SIP (cause 3 - No
> route to destination)
>
> Everyone is busy/congested at this time (1:0/0/1)
>
> Anyone faced that?
>
> Is it related to a paramater that control number of
> allowed channels per IP trunk? Maybe I have such
> parameters is 0 ? I do not know even if there is
such
> parameter.
>
> At the softswitch, I do not see even any attempt
> (nothing related to the dialed number), so why
> Asterisk does not send the called number to the
> softswitch and why asterisk assume there is not
> available channel?
>
> The softphone codec is g729a and the softswitch
> support such codec. Also, if it is a codec matter,
> then call should be send to the softswitch, and the
> softswitch will gives an error related to the codec
> missmatch.
>
> Any help?
>
> Regards
> Bilal Ghayad
__
Do You Yahoo!?
Tired of spam? Yahoo! Mail has the best spam protection around
http://mail.yahoo.com
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Re: [asterisk-users] Everyone is busy/congested: IP Trunk
On Fri, Oct 26, 2007 at 06:55:12AM -0700, bilal ghayyad wrote:
> Hi List;
Ip address to destination?
Unable to create channel of type SIP (cause 3 - No
route to destination)
i think you have the wrong ip information
>
> I established an SIP IP Trunk between Asterisk and
> another softswitch (asterisk registered on the
> softswitch successfully) and I saw this on the
> softswitch.
>
> >From firefly softphone, I was need to do a call to be
> via this softswitch (ofcourse, the softphone will send
> for asterisk and asterisk should route to the
> softswitch based on the extensions.conf
> configurations.
>
> But, always I receive this message (and the call does
> not even reach to the softswitch, it is not sended
> from Asterisk to the softswitch):
>
> Executing [EMAIL PROTECTED]:1]
> Dial("SIP/EgyptOeratorSIP-09f9bed0",
> "SIP/[EMAIL PROTECTED]") is new stack
>
> Unable to create channel of type SIP (cause 3 - No
> route to destination)
>
> Everyone is busy/congested at this time (1:0/0/1)
>
> Anyone faced that?
>
> Is it related to a paramater that control number of
> allowed channels per IP trunk? Maybe I have such
> parameters is 0 ? I do not know even if there is such
> parameter.
>
> At the softswitch, I do not see even any attempt
> (nothing related to the dialed number), so why
> Asterisk does not send the called number to the
> softswitch and why asterisk assume there is not
> available channel?
>
> The softphone codec is g729a and the softswitch
> support such codec. Also, if it is a codec matter,
> then call should be send to the softswitch, and the
> softswitch will gives an error related to the codec
> missmatch.
>
> Any help?
>
> Regards
> Bilal Ghayad
>
> __
> Do You Yahoo!?
> Tired of spam? Yahoo! Mail has the best spam protection around
> http://mail.yahoo.com
>
> ___
> --Bandwidth and Colocation Provided by http://www.api-digital.com--
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users
---end quoted text---
--
.-
Pablo Allietti
E-mail: [EMAIL PROTECTED] | LACNIC
Phone : +598 2 604 | http://LACNIC.NET
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